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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/test/PacketLossTest.h"
#include <cstdint>
#include <memory>
#include <string>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/neteq/default_neteq_factory.h"
#include "api/neteq/neteq.h"
#include "api/units/timestamp.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/EncodeDecodeTest.h"
#include "modules/audio_coding/test/RTPFile.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
ReceiverWithPacketLoss::ReceiverWithPacketLoss()
: loss_rate_(0),
burst_length_(1),
packet_counter_(0),
lost_packet_counter_(0),
burst_lost_counter_(burst_length_) {}
void ReceiverWithPacketLoss::Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
int file_num,
int loss_rate,
int burst_length) {
loss_rate_ = loss_rate;
burst_length_ = burst_length;
burst_lost_counter_ = burst_length_; // To prevent first packet gets lost.
StringBuilder ss;
ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
Receiver::Setup(neteq, rtpStream, ss.str(), channels, file_num);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
if (!_rtpStream->EndOfFile()) {
if (packet_counter_ == 0) {
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0) {
if (_rtpStream->EndOfFile()) {
packet_counter_ = 0;
return true;
} else {
return false;
}
}
}
if (!PacketLost()) {
_neteq->InsertPacket(
_rtpHeader,
ArrayView<const uint8_t>(_incomingPayload, _realPayloadSizeBytes),
Timestamp::Millis(_nextTime));
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
packet_counter_ = 0;
lost_packet_counter_ = 0;
}
}
return true;
}
bool ReceiverWithPacketLoss::PacketLost() {
if (burst_lost_counter_ < burst_length_) {
lost_packet_counter_++;
burst_lost_counter_++;
return true;
}
if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
lost_packet_counter_++;
burst_lost_counter_ = 1;
return true;
}
return false;
}
SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
void SenderWithFEC::Setup(const Environment& env,
AudioCodingModule* acm,
RTPStream* rtpStream,
absl::string_view in_file_name,
int payload_type,
SdpAudioFormat format,
int expected_loss_rate) {
Sender::Setup(env, acm, rtpStream, in_file_name, format.clockrate_hz,
payload_type, format);
EXPECT_TRUE(SetFEC(true));
EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
}
bool SenderWithFEC::SetFEC(bool enable_fec) {
bool success = false;
_acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
if (*enc && (*enc)->SetFec(enable_fec)) {
success = true;
}
});
return success;
}
bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
expected_loss_rate_ = expected_loss_rate;
return true;
}
return false;
}
PacketLossTest::PacketLossTest(int channels,
int expected_loss_rate,
int actual_loss_rate,
int burst_length)
: channels_(channels),
in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
: "audio_coding/teststereo32kHz"),
sample_rate_hz_(32000),
expected_loss_rate_(expected_loss_rate),
actual_loss_rate_(actual_loss_rate),
burst_length_(burst_length) {}
void PacketLossTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS
return;
#else
const Environment env = CreateEnvironment();
RTPFile rtpFile;
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
if (channels_ == 2) {
send_format.parameters = {{"stereo", "1"}};
}
std::string fileName =
test::TempFilename(test::OutputPath(), "packet_loss_test");
rtpFile.Open(fileName.c_str(), "wb+");
rtpFile.WriteHeader();
SenderWithFEC sender;
sender.Setup(env, acm.get(), &rtpFile, in_file_name_, 120, send_format,
expected_loss_rate_);
sender.Run();
sender.Teardown();
rtpFile.Close();
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
std::unique_ptr<NetEq> neteq = DefaultNetEqFactory().Create(
env, NetEq::Config(), CreateBuiltinAudioDecoderFactory());
ReceiverWithPacketLoss receiver;
receiver.Setup(neteq.get(), &rtpFile, "packetLoss_out", channels_, 15,
actual_loss_rate_, burst_length_);
receiver.Run();
receiver.Teardown();
rtpFile.Close();
#endif
}
} // namespace webrtc
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